Re: [Asterisk-Users] SIP call doesn't execute the 's'-extension
i have just started to configure access to the * over SIP-Phones. Therefore I have defined this SIP-Phone in sip.conf: [tobias] type=friend username=tobias secret=tobias auth=md5 host=dynamic reinvite=no dtmfmode=inband callerid=Tobias 1087006 allow=all context=javaAgi dtmfmode=rfc2833 As you can see i am directing calls from this user to the context [javaAgi] which is defined here in extension.conf: [javaAgi] exten = s,1,Answer() exten = s,2,Playback(code1000) exten = s,3,Hangup() exten = 1,1,Answer() exten = 1,2,Playback(code1000) exten = 1,3,Hangup() If i dial 1 on my SIP Phone everything works as suspected, the call is answered and the gsm-file is played. My understanding of the 's'-extension is, that it is executed then a call comes in an there is no extension wich matches the called number. But if i dial a random number i get an 404 Not found error. The s extension matches only when no digits are dialed. Dialing a 1 is a digit, so no match. Try playing around with exten=_.,1,Answer() and understand what the differences are. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP call doesn't execute the 's'-extension
Hi, Rich Adamson schrieb: The s extension matches only when no digits are dialed. Dialing a 1 is a digit, so no match. Oh, ok, i think now i understood. So the s extension is mainly the starting point for contexes which i reaches from other contexes, eg. because of a goto. When I receive a call there are naturally some digits dialed and with the pattern matching, you have suggested, i am able to react on them. Thanks a lot :- Tobias ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP call doesn't execute the 's'-extension
You only have a 1 in the javaAgi context and you aren't point the javaAgi to any other contexts, pressing anyting else but 1 will get a not found error because you only have 1 defined. If you want the call to continue you need to send it to another context or add more to the javaAgi context. Tobias Wolf wrote: Hi, i have just started to configure access to the * over SIP-Phones. Therefore I have defined this SIP-Phone in sip.conf: [tobias] type=friend username=tobias secret=tobias auth=md5 host=dynamic reinvite=no dtmfmode=inband callerid=Tobias 1087006 allow=all context=javaAgi dtmfmode=rfc2833 As you can see i am directing calls from this user to the context [javaAgi] which is defined here in extension.conf: [javaAgi] exten = s,1,Answer() exten = s,2,Playback(code1000) exten = s,3,Hangup() exten = 1,1,Answer() exten = 1,2,Playback(code1000) exten = 1,3,Hangup() If i dial 1 on my SIP Phone everything works as suspected, the call is answered and the gsm-file is played. My understanding of the 's'-extension is, that it is executed then a call comes in an there is no extension wich matches the called number. But if i dial a random number i get an 404 Not found error. Here is an snippet of what * tells me on sip debug, but i can't get a clue out of it: 12 headers, 13 lines Using latest request as basis request Sending to 10.3.4.98 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 10.3.4.98:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user 'tobias' Looking for 2 in javaAgi Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.3.4.98:5060;branch=z9hG4bK102F7CD4855F4C4E927C3398E3C57BF4 From: Tobias sip:[EMAIL PROTECTED];tag=2760968676 To: sip:[EMAIL PROTECTED];tag=as396962de Call-ID: [EMAIL PROTECTED] CSeq: 58303 INVITE User-Agent: evision PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Perhaps anyone can point me to the right direction ?? Tobias ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP call doesn't execute the 's'-extension
The s extension matches only when no digits are dialed. Dialing a 1 is a digit, so no match. Oh, ok, i think now i understood. So the s extension is mainly the starting point for contexes which i reaches from other contexes, eg. because of a goto. When I receive a call there are naturally some digits dialed and with the pattern matching, you have suggested, i am able to react on them. In the most general case, the exten=s is for incoming analog pstn lines (fxo ports) where the central office sends a call to your asterisk box by ringing the line. There are no digits sent to asterisk at all, therefore exten=s is used to handle that incoming call. Or, if you have an account with an itsp that sends incoming calls to your asterisk via iax/sip and doesn't send any digits to you, then exten=s is used for those as well. It really has nothing to do with 'other' contexts. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
